[..] |
ASTERISK-17000: File dir-welcome does not exist in any format |
ASTERISK-17001: SSRC is changing when DTMF sent |
ASTERISK-17002: saynumber(1,n) in Swedish doesn't work |
ASTERISK-17003: saynumber() fixes for Swedish |
ASTERISK-17004: Interger overflow in TIMEOUT(absolute) |
ASTERISK-17005: Deadlock on channel list during masquerade and channel search. |
ASTERISK-17006: [patch] Cannot forward voicemail with file storage backend |
ASTERISK-17007: Problem parked in asterisk |
ASTERISK-17008: Blind transfer one side issue |
ASTERISK-17009: [patch] Crash under FreeBSD 8.1 with 'core stop now' |
ASTERISK-17010: [patch] bad format in post_manager_event() i.e. ParkedCallGiveUp and ParkedCallTimeOut |
ASTERISK-17011: ParkedCall event is sent both as a status reply event and as a true event |
ASTERISK-17012: no VoicemailUserEntryComplete Event is sent when the voicemailuser list is empty |
ASTERISK-17013: [patch] SIP/TCP phones are not added to astdb - causes sip reload problems |
ASTERISK-17014: [patch] Missing P-Asserted-Identity |
ASTERISK-17015: [patch] const timeout = 30 makes app_originate pretty useless |
ASTERISK-17016: Conference with scheduler blocks on realtime ODBC connection to MSSQL database |
ASTERISK-17017: realtime queue wont update membername on queue show <queue> |
ASTERISK-17018: [patch] asterisk crash when dialing SIP/${var} where var is empty or not set |
ASTERISK-17019: Bug-Fix 13573 multi-view ldap content crash asterisk at suse linux with realtime extensions |
ASTERISK-17020: Possibility of using hashed password at jabber.conf |
ASTERISK-17021: Limitation in Playtones and indications.c does not allow ROH tone. |
ASTERISK-17022: compiling error in ubuntu 9.04 sparc |
ASTERISK-17023: cdr_mysql 1.4.11 does not log call answer or end times |
ASTERISK-17024: [patch] Forbidden - wrong password on authentication for REGISTER for '<username>' to '<sip>' |
ASTERISK-17025: [patch] Disable connected line updates for dahdi PRI channel |
ASTERISK-17026: attended transfer weird behaviour |
ASTERISK-17027: On session-timers refresh, after 422 response Asterisk set modified From/To/Contact on next re-INVITE |
ASTERISK-17028: [patch] install_prereq script needs subversion package on Debian |
ASTERISK-17029: [patch] one-way-audio when chanspy activated |
ASTERISK-17030: Its not clear that enabling more verbosity in the contributed default file makes asterisk NOT start as a daemon |
ASTERISK-17031: [patch] main/asterisk.c compile errors |
ASTERISK-17032: ast_expr2.y undefined references with DEBUG_FD_LEAKS |
ASTERISK-17033: Blind transfer not working from DAHDI to SIP-to-SIP |
ASTERISK-17034: [regression] Can't receive SIP INFO DTMF when using Read() without Answer() |
ASTERISK-17035: DTMF RFC2833 Not recognized |
ASTERISK-17036: [patch] Output of queue_log stopped after several hours |
ASTERISK-17037: Cross compile time if the version is equal to and greater than r284478 will hang |
ASTERISK-17038: [patch] Mixmonitor does not parse file path proper if it contain a . (period) |
ASTERISK-17039: meetme "a" option not working properly with realtime |
ASTERISK-17040: C should not receive request call again after C cancel if B blind transfer using atxfer call C |
ASTERISK-17041: B atxfer call C, B retrieve call after C cancel call, A <-> B one way voice |
ASTERISK-17042: IAX2 CODEC_PRES wrong (offset error?) |
ASTERISK-17043: [patch] segfault with 'core stop gracefully' |
ASTERISK-17044: Call torn down upon connection when early media 183 used |
ASTERISK-17045: [patch] chan_ooh323 cannot register to Tandberg Gatekeeper |
ASTERISK-17046: [patch] Deadlock on SIP blind transfer (REFER) |
ASTERISK-17047: asterisk will not load under Mac OS 10.4 (Tiger) |
ASTERISK-17048: Convert wav16 file to sln16, bad quality |
ASTERISK-17049: [patch] Setting linear queue strategy requires asterisk restart |
ASTERISK-17050: [patch] 'usegmtime=yes' is ignored and CDR(end) is not inserted |
ASTERISK-17051: [patch] When a call going out an NT-PTMP port gets rejected, Asterisk crashes |
ASTERISK-17052: False protocol error on waiting call |
ASTERISK-17053: meetme 'c' option is ignored if 'q' is also used - these used to be separate. |
ASTERISK-17054: Exchange AutoAttendant cannot make calls back to asterisk (worked in 1.6.2.2 + 1.6.2.6) |
ASTERISK-17055: [patch] outbound google voice calls fail (staring approx 30Nov2010) |
ASTERISK-17056: [patch] Using static extensions.conf with res_config_curl segfaults on startup |
ASTERISK-17057: [patch] Nokia disconnects (error 104), missing SMSSRC and unicode SMS support |
ASTERISK-17058: [patch] Asterisk 1.8.1-rc1 crashes in cdr.c line 1201 after a parked call catched with parkedcall() is hungup |
ASTERISK-17059: [patch] Improvements to install_prereq |
ASTERISK-17060: rtp.c memory leak |
ASTERISK-17061: Wait for leader with Music On Hold allows crosstalk between participants |
ASTERISK-17062: IAX2 bindaddr doesn't bind to IP address |
ASTERISK-17063: Meetme doesn't detect peer unreachable |
ASTERISK-17064: Segmentation Fault with flag LOW_MEMORY |
ASTERISK-17065: [patch] Device state providers: Chicken and egg problem - initialization of extension state |
ASTERISK-17067: Long lines in call files cause spurious syntax error |
ASTERISK-17068: ${CHANNEL} broken in 'failed' extension |
ASTERISK-17069: Callfile retries behave erratically as file size grows |
ASTERISK-17070: SIP response |
ASTERISK-17071: Realtime for ODBC & Mysql is diffrent |
ASTERISK-17072: [patch] Segfault when ExternalIVR() app. immediately sends S command due to race condition |
ASTERISK-17073: Regression - T.38 no longer functions |
ASTERISK-17074: No audio on incoming calls when callerid (database cidname) was saved with a newline |
ASTERISK-17075: [patch] SIP calls to H323 Tandberg Endpoints are not fully answered |
ASTERISK-17076: Asterisk 1.8.0 locks up under OpenSUSE 10.3 |
ASTERISK-17077: SIP response don't worked $(HASH(SIP_CAUSE,<slave-channel-name>)} |
ASTERISK-17078: "transmit_refer" builds REFER with no host in case of anonymous call transfer |
ASTERISK-17079: Ealry media is lost on accepted calls (with open r2 support, Elastix 2 flavor) |
ASTERISK-17080: [patch] Deadlock in chan_sip |
ASTERISK-17081: [patch] Error building netsock library on OpenSolaris 5.11 snv_134b |
ASTERISK-17082: [patch] doc/tex dir removed, but corresponding entries in Makefile & README still exists |
ASTERISK-17083: [patch] leak in meetme |
ASTERISK-17084: Realtime expires IAX2 registration wrongly |
ASTERISK-17085: [patch] Configuring settings in /etc/defaults/asterisk cause the init.d process to stay on STDOUT |
ASTERISK-17086: timeout parameter always honoured when no response to signalling |
ASTERISK-17087: ./configure: syntax error at line 5524: `GTK2_INCLUDE=$' unexpected |
ASTERISK-17088: [fr] Female gender missing for digit "one" in holdtime announcement |
ASTERISK-17089: Dialplan does not work as it should |
ASTERISK-17090: bindaddr=[::]:5060 does not work as expected under FreeBSD |
ASTERISK-17091: Problem on TRANSFER using SNOM transfer Button |
ASTERISK-17092: [patch] moh reload leaks file descriptors to dahdi timing channel |
ASTERISK-17093: Crash with malformed (invalid) exten in extensions.conf |
ASTERISK-17094: netsock2.c: getaddrinfo("", "(null)", ...): ai_family not supported |
ASTERISK-17095: [patch] CDR(billsec) is not inserted properly for unanswered calls |
ASTERISK-17096: [patch] cel.conf mentions USER_EVENT when it should be USER_DEFINED |
ASTERISK-17097: [patch] CEL PostgreSQL doesn't include logging for 'eventextra' |
ASTERISK-17098: [patch] IPv6 address truncated when stored in realtime |
ASTERISK-17099: [patch] Mandriva support for install_prereq script |
ASTERISK-17100: forkCDR - variables mismatch |
ASTERISK-17101: SIP crash on transfer |
ASTERISK-17102: 1.4.38 does not write external callerID number into SIP From: header |
ASTERISK-17103: Unable to establish SRTP if receive INVITE with no SDP |
ASTERISK-17104: [patch] Uri encoded Refer-To fails to match callid, attended transfer fails |
ASTERISK-17105: call from autocreated peer forces coredump |
ASTERISK-17106: [patch] Schema selection support |
ASTERISK-17107: [patch] "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 |
ASTERISK-17108: call files: Calls get stalled during stream_file, asterisk "hangs up" without hanging up |
ASTERISK-17109: Codec negotiation algorithm |
ASTERISK-17110: MySQL Realtime - musicclass/musiconhold ignored |
ASTERISK-17111: [patch] Ldap Can't modify unknown attribute |
ASTERISK-17112: [patch] Don't answer the incoming call until ready to bridge |
ASTERISK-17113: [patch] Contexts with a 'switch' statement in a 's' extension adds MSet to the beginning of the Context |
ASTERISK-17114: [patch] *** glibc detected *** /usr/sbin/asterisk: free(): invalid pointer on shutdown |
ASTERISK-17115: Undefined SIP users can exploit default context to make calls |
ASTERISK-17116: Random crash when Asterisk is started |
ASTERISK-17117: [patch] IAX2 Retry Time Review |
ASTERISK-17118: [patch] Voicemail files out of sequence |
ASTERISK-17119: Deadlock after receiving calls generated with sipp |
ASTERISK-17120: Asterisk does not end call properly and stops reacting to following SETUP messages |
ASTERISK-17121: Problems with Escaping Characters on SIP Calls in Dial() and Transfer() Applications |
ASTERISK-17122: sip.conf example files contains wrong example |
ASTERISK-17123: [patch] fix SIP indicate deadlocks when lots of state changes |
ASTERISK-17124: Asterisk does not hangup a channel after endpoint hangs up when using FastAGI |
ASTERISK-17125: [patch] Sqlite3 requires -lpthread to build in configure.ac |
ASTERISK-17126: [patch] Random Deadlocks in <chan_sip.c> or <channel.c> !?! |
ASTERISK-17127: Segmentation fault when using dynamic hints and func_odbc |
ASTERISK-17128: deadlock on 'core stop gracefully' |
ASTERISK-17129: [patch] Asterisk Crashing/Hangs |
ASTERISK-17130: [patch] [regression] Resequencing of mailbox not working as expected. |
ASTERISK-17131: [patch] Fallthrough in get_member_status can cause unwarranted exit from queue |
ASTERISK-17132: call queue member noanswer but not record cdr with no answer. |
ASTERISK-17133: [patch] minivm: when sending mail and using volgain |
ASTERISK-17134: [patch] Schema selection support for cel_odbc.conf |
ASTERISK-17135: SIP autocreated peers crash asterisk on call |
ASTERISK-17136: SPEECH_ENGINE should be readable |
ASTERISK-17137: [patch] New manager Action: QueueSync |
ASTERISK-17138: [patch] Asterisk not re-registering after it receives "Forbidden - wrong password on authentication" |
ASTERISK-17139: [PATCH] Adds astsounddir configuration option. |
ASTERISK-17140: [patch] chan_dahdi inserts empty COLP |
ASTERISK-17141: [patch] Sending out unnecessary PROCEEDING messages breaks overlap dialing |
ASTERISK-17142: Asterisk 1.4.38, 1.6.2.15 and 1.8.1 with debug flags crashes on load test |
ASTERISK-17143: [patch] Asterisk hangs with no calls being able to go in or out, and the console does not respond to commands |
ASTERISK-17144: The function ast_rtp_set_rtpmap_type_rate returns a wrong result. |
ASTERISK-17145: IAX2 REGAUTH Failure dependent on Trunk Name |
ASTERISK-17146: [patch] Problem with dialing SIP peer that is not reachable. |
ASTERISK-17147: [patch] Asterisk produces many zombie processes while under load. |
ASTERISK-17148: [patch] SIP REFER transfers do not work |
ASTERISK-17149: When caller sent to queue, MoH plays for a quick second then ringback tone |
ASTERISK-17150: hint for confbridge fails |
ASTERISK-17151: Deadlock - Asterisk stops processing sip calls after a few calls. |
ASTERISK-17152: Empty Redirection Number in outgoing SETUP when calling from IAX |
ASTERISK-17153: [patch] Memory leaking in calendars |
ASTERISK-17154: CDR fileds not being written from "h" extension or after "Dial" command completes. |
ASTERISK-17155: qualifyfreq not respected when user is "UNREACHABLE" |
ASTERISK-17156: [patch] The default Polycom directory (000000000000.cfg) don't fit in small 330 and 331 phones |
ASTERISK-17157: [patch] ast_update2_realtime fails to update multiple columns with MySQL driver |
ASTERISK-17158: [patch] res_config_mysql.so crashes when processing "#include" with a file which doesn't exist |
ASTERISK-17159: [patch] Allow more control over the output of pri debug |
ASTERISK-17160: [patch] Demonstration for all the features of res_phoneprov |
ASTERISK-17161: [patch] Note more settings in users.conf.sample to guide the user |
ASTERISK-17162: make of chan_ooh323 fails "ret undeclared" |
ASTERISK-17163: Asterisk consumes 100% of CPU on Mac OS X |
ASTERISK-17164: [patch] if fastStart is enabled, Progress message is sent when it shouldn't |
ASTERISK-17165: app_queue deadlocks if weight is set |
ASTERISK-17166: Missing MySQL table defintion for extensions |
ASTERISK-17167: [patch] MySQL table definitions are broken (wrong commas and a wrong name) |
ASTERISK-17168: Extension not matched. |
ASTERISK-17169: [regression] iax auth rsa failed with policie not found |
ASTERISK-17170: Problem with unistim on Asterisk 1.8.1.1 |
ASTERISK-17171: [patch] Each time a device sends a REGISTER, the error "No address associated with nodename" appears |
ASTERISK-17172: [patch] OOH323 Incoming and Outgoing Calls Fail Avaya with Asterisk 1.8.1.1 |
ASTERISK-17173: Asterisk Crash possibly with app_swift |
ASTERISK-17174: [patch] On Darwin, pbx_spool stops looking for work after 20-30 minutes of idle time |
ASTERISK-17175: [patch] IAX Incorrectly reports IAX/Registry astdb host:port invalid |
ASTERISK-17176: [patch] WARNING message for each IAX peer with a qualifyfreqnotok setting |
ASTERISK-17177: [patch] main/xmldoc.c error message is backwards -- fix that, and remove the sources of the error while we're at it |
ASTERISK-17178: CHANNEL variable is set incorrectly by Asterisk |
ASTERISK-17179: [patch] IMS TEL URI incoming INVITE RFC 3966 not recognized |
ASTERISK-17180: [patch] [new feature] support for T.38 pass-through |
ASTERISK-17181: Unwanted Hangup On dial |
ASTERISK-17182: [patch] Not correct encode Header of mail message |
ASTERISK-17183: Multiple Parking Lots Being Redirected to the Wrong Parking Lot |
ASTERISK-17184: SIP channel tried to release unowned mutex in handle_incoming() |
ASTERISK-17185: [patch] SIP CHANNEL(rtpqos,audio,...) variables missing. |
ASTERISK-17186: [patch] Up to 6 [DEBUG] messages forced to console on every call |
ASTERISK-17187: [patch] No timeout on T.38 re-INVITE |
ASTERISK-17188: [patch] p->chan can disappear between test and lock in deadlock avoidance in local_hangup |
ASTERISK-17189: [patch] CEL logging to ODBC Eventtype error on userdefined type |
ASTERISK-17190: [patch] CEL logging to ODBC field not being stored. |
ASTERISK-17191: [patch] p->owner unregistered from module prematurely in local_hangup. |
ASTERISK-17192: get_header doesn't pick the right header when compact form is mixed with non-compact form |
ASTERISK-17193: ChanSpy leaves channel in Up state behind, warnings |
ASTERISK-17194: Asterisk Remote Connection blocked while using with incoming IAX Fax (iaxmodem 2.x) and hylafax |
ASTERISK-17195: Voicemail MWI - deleted messages are not notified to the extension |
ASTERISK-17196: High CPU Usage / Choppy Sound (res_timing_pthread) |
ASTERISK-17197: segmentation fault |
ASTERISK-17198: Forwarding voicemail generate error in multi-tenant configuration |
ASTERISK-17199: Asterisk may crash when PostgreSQL database is reconnecting |
ASTERISK-17200: handle_request_info uses uninitialized string buffer |
ASTERISK-17201: [patch] Create a manager event when the app_dial creates a new call with a new unique id |
ASTERISK-17202: [patch] Required support for DTMF-R2 on chan_dahdi |
ASTERISK-17203: [patch] Adjusted source comment to real situation about inheritance of a category or template in config files |
ASTERISK-17204: Unable to set outbound call CODEC using SIP_CODEC_OUTBOUND |
ASTERISK-17205: [patch] Using 'asterisk -r' no longer displays welcome_message |
ASTERISK-17206: [patch] Allow external commands to send mailbox refreshes |
ASTERISK-17207: Problem reading a message which was previously saved as unread in INBOX. |
ASTERISK-17208: [patch] Read func CHANNEL() on sip channel without arg will crash asterisk |
ASTERISK-17209: [patch] Codec negotiation fails on IAX calls from 1.8.1.1 to 1.8.1.1 |
ASTERISK-17210: The response to a CANCEL request does not always comply with RFC3261 |
ASTERISK-17211: [patch] AMI redirect from meetme - calls fail |
ASTERISK-17212: [patch] Indicate log level argument for Log() is not optional |
ASTERISK-17213: Channel variables apparently not available as application arguments in features.conf application map |
ASTERISK-17214: DNS Lookup blocking registration |
ASTERISK-17215: [patch] [regression] asterisk 1.8 tarball sound files ignored, Makefile re-downloads them. |
ASTERISK-17216: Asterisk tarballs missing .sha1 checksum files |
ASTERISK-17217: [patch] race condition in setting local capabilities for Setup when fastStart is enabled |
ASTERISK-17218: spooler stop after some time |
ASTERISK-17219: RTCP conflict avoidance not handled |
ASTERISK-17220: Text Correction |
ASTERISK-17221: Asterisk SVN 1.8 running at 99% CPU |
ASTERISK-17222: [patch] Realtime Peers Cannot Register |
ASTERISK-17223: [regression] Snom BLF subscription does not work |
ASTERISK-17224: Manager Event Interface w/Digest authentication does not work! |
ASTERISK-17225: [patch] Can't call between Tandberg MPS 200 video bridge and SIP softphone on Asterisk running chan_ooh323 |
ASTERISK-17226: [patch] r299990 broke file paths on Linux |
ASTERISK-17227: No RTP port update when SIP RE-INVITE is received |
ASTERISK-17228: Dial MulticastRTP channels with A option kills asterisk |
ASTERISK-17229: [patch] Get Real Channel of a Dahdi Call |
ASTERISK-17230: [patch] [regression] Crash with "realtime show pgsql cache" |
ASTERISK-17231: [patch] unopenable spool files not deleted |
ASTERISK-17232: [patch] Static entry for Polycom 331 split firmware |
ASTERISK-17233: DTMF tones not recognized with Exchange 2010 Unified Messaging |
ASTERISK-17234: Asterisk 1.8.1.1 "new" sip peers status. |
ASTERISK-17235: hint for confbridge fails |
ASTERISK-17236: Hangup on communication stage due to GSM or iLBC codec. |
ASTERISK-17237: inband DTMF cannot be detected and trigger service execute when A and B both use u-law (the same codec) |
ASTERISK-17238: Redirect channels FROM Meetme causes hangup |
ASTERISK-17239: Local channel: variables do not propagate on masquerade until Wait() |
ASTERISK-17240: [regression] Local channel: Caller ID in .call file ignored |
ASTERISK-17241: Format list should be copied before being iterated in ast_filehelper |
ASTERISK-17242: ${CALLERID(subaddr)} returns the trailing '}', ${CALLERID(dnid-subaddr)} works OK |
ASTERISK-17243: Cannot originate DAHDI PRI calls after 1.8.1.1 upgrade |
ASTERISK-17244: atxfer doesn't work |
ASTERISK-17245: Asterisk deadlocks with no errors |
ASTERISK-17246: How to retrive recorded call using your API (in Test Suite) |
ASTERISK-17247: Asterisk losing timezone if /etc/localtime is changed |
ASTERISK-17248: [patch] Remove reference to "priorityjumping" configuration option in extensions.conf.sample |
ASTERISK-17249: SIP registration message sequencing issue causes inability to register |
ASTERISK-17250: [patch] registration rejected - not a local domain |
ASTERISK-17251: asterisk sometimes returns 0 on segfault, safe_asterisk doesn't restart |
ASTERISK-17252: Error to compile Asterisk 1.8.2 |
ASTERISK-17253: sip call fails to hang up - asterisk uses 99% resources |
ASTERISK-17254: Dial MulticastRTP channel with A option can't play the file |
ASTERISK-17255: Possible deadlock on 1.6.2.12, 14 and 15 |
ASTERISK-17256: Ability to specify path to self signed SSL cert CA |
ASTERISK-17257: Asterisk crashes on "dialplan reload" |
ASTERISK-17258: missing Contact header in 200 OK to INVITE |
ASTERISK-17259: asterisk does not respond with ACK on retransmission of 200 OK after it sent ACK |
ASTERISK-17260: PGSQL db update issue |
ASTERISK-17261: No Re-Invite message from Asterisk to the peer when trying for FAX pass-through |
ASTERISK-17262: Memory corruption crash in realtime voicemail using res_config_mysql when attempting to forward message using directory |
ASTERISK-17263: [patch] No MOH on Call Park and Exceptionally long voice queue length ... |
ASTERISK-17264: [patch] [regression] Call Pickup Hangs Asterisk (deadlock?) |
ASTERISK-17265: Asterisk as a SIP proxy |
ASTERISK-17266: codec_dahdi.c:147 lintoulaw: Out of buffer space! |
ASTERISK-17267: SIP channel not hung up on BYE |
ASTERISK-17268: [patch] [regression] crash while looking up realtime presence hint |
ASTERISK-17269: [patch] MeetMe-like 'x' option for ConfBridge |
ASTERISK-17270: Asterisk 1.8 branch failed to compile |
ASTERISK-17271: DAHDI channel becomes busy/unusable after it's called and not picked up |
ASTERISK-17272: Incoming calls stop working after changing IP in DNS SRV |
ASTERISK-17273: [patch] - Incorrect address specified in SIP re-INVITE with T.38 when directmedia enabled |
ASTERISK-17274: Monitor() channel mixing randomly stops working |
ASTERISK-17275: Problems with queue when there is no answer app before queue app |
ASTERISK-17276: [patch] Chan_dahdi hangs up after first ring when doing dtmf cid detection without polarity reversal |
ASTERISK-17277: chan_iax2 - timing interface missing |
ASTERISK-17278: [patch] chan_h323.c:977 __oh323_rtp_create: Unable to create RTP session: Address family not supported by protocol |
ASTERISK-17279: [patch] Command 'module load pbx_lua.so' failed |
ASTERISK-17280: MOH on park stops and restarts - causes CDR error as well |
ASTERISK-17281: when using chan_gtalk the tos bit set in either the sip.conf and or the iax,conf is ignored |
ASTERISK-17282: [patch] Unable to choose which SRTP suite to offer |
ASTERISK-17283: ChannelRedirect hanging up a channel who is in a ChanSpy |
ASTERISK-17284: T38 passthrough doesn't work in glare reinvite situation |
ASTERISK-17285: [regression] Transfer from queue agent does not change his state |
ASTERISK-17286: [patch] Cdr_syslog loses configuration on reload |
ASTERISK-17287: Random deadlocks in channel.c line 2744 (__ast_read), Asterisk freeze |
ASTERISK-17288: [regression] Asterisk 1.8x, SIP 484 set HANGUPCAUSE to 111 instead of 28 |
ASTERISK-17289: When using u and g in dial if the call is bridged there is no audio once called hangs up |
ASTERISK-17290: [REGRESSION]: Attended SIP transfer to park extension segfaults Asterisk |
ASTERISK-17291: [REGRESSION]: Transfer to park extension leaves zombie channels if incoming PRI channel disconnects and sometimes segfaults * |
ASTERISK-17292: [REGRESSION]: Files based voicemail message re-sequencing still not working as expected. |
ASTERISK-17293: [patch] Wrong country code identifier "%2B" instead of "+" in Remote-Party-ID |
ASTERISK-17294: [patch] high iowait due to ast_db_put of realtime peers |
ASTERISK-17295: Call Transfer/ Call Forwarding not working |
ASTERISK-17296: Calls didn't appear in CDR |
ASTERISK-17297: sip deadlock with explanation |
ASTERISK-17298: Call drops |
ASTERISK-17299: [patch] Compile Error - odbc_storage enabled |
ASTERISK-17300: [patch] Fax with T.38 enabled doesn't work over ooh323 |
ASTERISK-17301: Memory Leak in chan_sip.c with Asterisk Version 1.8.2.x & 1.8.3.x |
ASTERISK-17302: Segfault in chan_iax2 |
ASTERISK-17303: Asterisk doesn't respond ACK to 200 OK retransmission |
ASTERISK-17304: [patch] [regression] rtpkeepalive no longer works, but is still documented to work |
ASTERISK-17305: Caller name does not preserve "(" parenthese in the from field |
ASTERISK-17306: SIP Realtime: Peer isn't deleted/cleared from 'sip show peers' when the phone unregister |
ASTERISK-17307: ConfBridge mute option 'm' fails under a specific scenario |
ASTERISK-17308: no Connected Line Presentation (COLP) transparency for SIP to SIP calls via Asterisk |
ASTERISK-17309: Sometime Server Crashes when receiveing/calling through chan_mobile |
ASTERISK-17310: [patch] Alignment issue cause multiple failures with 1.8 on ARM |
ASTERISK-17311: "Require: timer" header still being sent |
ASTERISK-17312: Add support for NetBSD/Solaris style atomic operations. |
ASTERISK-17313: CallerID not update on Outgoing(E1) Attended Xfer |
ASTERISK-17314: Casting alignment problem in send_client causes bad seq to be generated on ARM platform |
ASTERISK-17315: sip_xmit warning |
ASTERISK-17316: On cdr table Caller number in not being stored in clid and src, instead extension number is on outgoing calls. |
ASTERISK-17317: Cannot park a call more than once |
ASTERISK-17318: Crash when reloading dialplan with syntax error |
ASTERISK-17319: Asterisk Loses gateway and chansip stops |
ASTERISK-17320: [patch] Add support for snom phones |
ASTERISK-17321: rev. 305040: asterisk is not initialized because the g729a-codec is crashing |
ASTERISK-17322: persian say number feature |
ASTERISK-17323: manager_park can deadlock with ast_channel_free, for channel 1 of the park operation (channel list v channel lock) |
ASTERISK-17324: [regression] Implicit declaration of copy yields broken app_voicemail when building with IMAP support |
ASTERISK-17325: Asterisk segfaults after a number of unsuccessfull CDR submits |
ASTERISK-17326: Crash of Asterisk: deprecated BYE/Also transfer method |
ASTERISK-17327: UA2 not receiving re-INVITE after Asterisk redirect action |
ASTERISK-17328: When using ugc in dial - If first leg hangs up call is in limbo |
ASTERISK-17329: [branch] DTMF on outbound call leg does not match inbound DTMF duration |
ASTERISK-17330: Russian Sounds |
ASTERISK-17331: no musiconhold audio if no other sound has been played earlier during incoming call |
ASTERISK-17332: crash when transport not set |
ASTERISK-17333: [patch] Use of Google XMPP Extensions for Google Talk/Voice |
ASTERISK-17334: [patch] [regression] Dial() and Queue() with a macro argument are broken by AEL macro compilation change |
ASTERISK-17335: [path] func_env.c FILE estatement left file descriptors open |
ASTERISK-17336: Segfault when transferring call |
ASTERISK-17337: [regression] When caller sent to queue, MoH stops, only ringback tone |
ASTERISK-17338: Combination dtmfmode=info, directmedia=yes/update and transfers blocks Asterisk. |
ASTERISK-17339: [patch] memory leak with IAX in 1.8 |
ASTERISK-17340: [patch] Out-of-dialog MWI from a SIP Trunk in asterisk 1.8.2.2 always responds with 489 Bad event |
ASTERISK-17341: [patch] MWI subscription to remote server crashes during parse |
ASTERISK-17342: [review][branch] DTMF is delayed through features.c |
ASTERISK-17343: wrapuptime doesn't work |
ASTERISK-17344: Voicemail files out of sequence (voicemail message re-sequencing not always working correctly) |
ASTERISK-17345: SIP CHANNEL(rtpqos,audio,...) variables missing |
ASTERISK-17346: [patch] MIXMON_ARGS not processed when call being monitored via chanspy |
ASTERISK-17347: [REGRESSION] Heavy locking unlocking on incoming PRI channel |
ASTERISK-17348: audiohook.c:224 audiohook_read_frame_both: Read factory 0x9759380 and write factory 0x9759d9c both fail to provide 160 samples |
ASTERISK-17349: calls not routing to polycom handset after transfer using softkeys |
ASTERISK-17350: Program terminated with signal 11, Segmentation fault |
ASTERISK-17351: [regression] changeset 298596 harm the ring/moh logic in queues |
ASTERISK-17352: [patch] Deadlock of SIP takes out server |
ASTERISK-17353: The MixMonitor application fails to execute the "command" upon termination if ExtenSpy is used on the channel |
ASTERISK-17354: Dynamic prioritization of agents in a queue based on an outside metric |
ASTERISK-17355: Asterisk lockup on reload |
ASTERISK-17356: Asterisk Crash while starting |
ASTERISK-17357: In Red Hat 6.0 it locks uo |
ASTERISK-17358: SIP RTP with 2 UA and Asterisk all NATTED through a stateful (but not SIP aware) firewall |
ASTERISK-17359: [patch] astobj2.c does not report ref changes during internal_ao2_callback for refcounter util |
ASTERISK-17360: [patch] Null values in format strings lead to segfault |
ASTERISK-17361: SoftHangup not effective on 1.8.2.3 |
ASTERISK-17362: Unsolicited SIP notifies cannot be routed to MWI lights on individual phones |
ASTERISK-17363: [patch] CallCompletionRequest() / Cancel() exit non-zero if no call/pending call |
ASTERISK-17364: [regression] Incoming DTMF on PRI channel is mis-interpreted causing logins to voicemail main to fail authentication |
ASTERISK-17365: test |
ASTERISK-17366: [patch] Skill routing |
ASTERISK-17367: [patch] AGI streamfile turns off moh, doesn't turn it on |
ASTERISK-17368: [patch] App voicemail inconsistencies with file operations when using IMAP_STORAGE |
ASTERISK-17369: CALLCOMPLETION(cc_monitor_policy) has no affect after the call attempt ends |
ASTERISK-17370: [patch] FD 32767 exceeds the maximum size of ast_fdset |
ASTERISK-17371: Segmentation fault in __ast_module_user_remove () |
ASTERISK-17372: [patch] Senddtmf(f) fails with don't know how to indicate condition 9 when using SIP |
ASTERISK-17373: [patch] asterisk -rx 'core show version' returns an ANSI string, which is not evaluated correctly with debian init.d script. |
ASTERISK-17374: [patch] Enhancement to allow retreival from specific parking lot |
ASTERISK-17375: Mute/Unmute lock/unlock messages are not being played when activated via AMI |
ASTERISK-17376: Streaming audio file through Local channel to a few SIP devices randomly loses audio |
ASTERISK-17377: asterisk -rx 'core show version' returns an ANSI string, may not evaluated correctly with various dists init.d scripts |
ASTERISK-17378: [patch] SEGFAULT in remote_bridge_loop after a SIP to SIP attended transfer with external IAX2 or DAHDI call |
ASTERISK-17379: bfin-uclinux-uclibc:TLS extension can not make calls |
ASTERISK-17380: [patch] Segmentation fault - SVN-branch-1.8-r307314 |
ASTERISK-17381: Local channel DTMF detection broken by 17370 |
ASTERISK-17382: [patch] Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.) |
ASTERISK-17383: When locking a conference using the AMI or console, the conf-lockednow message is not played to the users |
ASTERISK-17384: [patch] Security issue in originate, system permission bypassed if using async |
ASTERISK-17385: [patch] Add Device State Information CCSS for Generic Devices |
ASTERISK-17386: [patch] res_config_ldap with malloc_debug produces munmap_chunk(): invalid pointer: |
ASTERISK-17387: [patch] Deadlock In chan_sip (conlock / cb_extensionstate |
ASTERISK-17388: [patch] Deadlock sip_read check_rtp_timeout #16608 |
ASTERISK-17389: Asterisk hangs and takes 100% cpu usage |
ASTERISK-17390: Low soundquality when 3 users use G722 codec and one user Ulaw codec |
ASTERISK-17391: [patch] Allow a per agent ringinuse setting [ignorebusy] |
ASTERISK-17392: [patch] func_odbc insertsql option may not work |
ASTERISK-17393: RTP Early Media not Passed to Caller |
ASTERISK-17394: Call drop with chan_sip |
ASTERISK-17395: GTalk Channel Naming Deviates from TECH/name-SESSION |
ASTERISK-17396: Res_musiconhold Kills Mpg123 Upon Initial Start or Reload if a Streaming Source Defined |
ASTERISK-17397: features reload does not clear old configuration and always include 700 on parkedcalls |
ASTERISK-17398: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) |
ASTERISK-17399: deadlock related to "handle_tcptls_connection" - tcp transport |
ASTERISK-17400: [patch] session-timers=refuse is not enforced for *caller* |
ASTERISK-17401: DTMF CallerID polarity reversal detection |
ASTERISK-17402: No error, if meetme conference number is not in meetme.conf |
ASTERISK-17403: VERY IMPORTANT: Asterisk hangup the line if the customer presses any digit 6 or more times |
ASTERISK-17404: Asterisk detect hangup 10 seconds after line has been hangup |
ASTERISK-17405: [patch] ACK is ignored upon call-pickup with sendrpid=yes |
ASTERISK-17406: [patch] app_meetme volume adjustment menu options error |
ASTERISK-17407: [patch] Seems Like ast_read / timerfd_timer_ack Causes lockups and resource drain |
ASTERISK-17408: [patch] IAX2 incompatible requested/capability between 1.8 SVN and any 1.8.x.y tagged release <= 1.8.2.3 |
ASTERISK-17409: [patch] Session timer refresher incorrect |
ASTERISK-17410: Video dynamic RTP payload type negotiation incorrect when directmedia enabled |
ASTERISK-17411: [patch] func_odbc insertsql option may not work on SQL_NODATA |
ASTERISK-17412: Asterisk Hangs/ CPU Usages goes up to 500% |
ASTERISK-17413: [patch] MONITOR_FILENAME should be MIXMONITOR_FILENAME in documentation of MONITOR_EXEC |
ASTERISK-17414: [patch] Crashing when using local channels and realtime on asterisk 1.8.3-rc2 |
ASTERISK-17415: random deadlock |
ASTERISK-17416: [patch] Jabber and Gtalk Do Not Load After the Reboot |
ASTERISK-17417: [patch] fix for crash in strcompare_l in sip/reqresp_parser.c - possible null c_locale |
ASTERISK-17418: Asterisk deadlocks with a SIP channel locked |
ASTERISK-17419: Asterisk 1.4.29.1: Quiet mode doesn't work in dyanmic MeetMe |
ASTERISK-17420: [patch] Add a group to the channel on answer of member channel |
ASTERISK-17421: [patch] Proposed better handling of negative penalty's |
ASTERISK-17422: Unable to Authenticate() against AstDB |
ASTERISK-17423: [regression] contrib/scripts/meetme.sql doesn't contain all fields |
ASTERISK-17424: [patch] ParkedCall() does not update Connected Line information |
ASTERISK-17425: [patch] Run a macro on picked up channel when bridging |
ASTERISK-17426: default subscribecontext ignored when allowguest=yes |
ASTERISK-17427: SIP over TCP and TLS does not appear to support NAT=yes |
ASTERISK-17428: [patch] Allow "Comedian Mail" branding to be removed |
ASTERISK-17429: [patch] Allows a user to dynamically forward a voicemail or to leave a voicemail to more that one mailbox |
ASTERISK-17430: ParkedCall() with no extension should pickup first available call and does not |
ASTERISK-17431: [patch] Deadlock with attended transfer of SIP call |
ASTERISK-17432: ChanSpy causes hangup of target channel |
ASTERISK-17433: A voicemail password that starts with a '*' results in a invalid mailbox |
ASTERISK-17434: Crash on chan local |
ASTERISK-17435: Agent and hint status "On Hold" messed up |
ASTERISK-17436: random deadlocks - SIP messages not being processed |
ASTERISK-17437: C# API For asterisk version 1.8.2.3 |
ASTERISK-17438: [patch] TCP TLS session open failure dumps core |
ASTERISK-17439: Crash in 'core show locks' |
ASTERISK-17440: Dialed Number Truncated in the "Dialing..." Output on the Console |
ASTERISK-17441: Crash in generic_http_callback |
ASTERISK-17442: [patch] Canary failure |
ASTERISK-17443: [patch] MixMonitor does not record FollowMe calls when "on bridge" flag set |
ASTERISK-17444: callfiles stops after sometime |
ASTERISK-17445: [patch] ACK after INVITE-481 not acecpted |
ASTERISK-17446: Asterisk hangs when generating SIP calls with sipp |
ASTERISK-17447: Realtime field 'fullcontact' populated with invalid data |
ASTERISK-17448: [patch] Segmentation fault in strlen () from /lib64/libc.so.6 |
ASTERISK-17449: SMS is readed correctly but the carrier keep sending the same |
ASTERISK-17450: busydetect not detecting some patterns |
ASTERISK-17451: [patch] Odd Behavior when dialed sip channel doesn't exist |
ASTERISK-17452: parking_offset is write-only in features.conf |
ASTERISK-17453: Free parking slot logic broken if randomising start - stops early and doesn't recognize it failed (#16428 is not complete fix) |
ASTERISK-17454: [patch] support gmime-2.4 |
ASTERISK-17455: The "Unlink" bridge event in channel.c |
ASTERISK-17456: crash after attended transfer |
ASTERISK-17457: Asterisk 1.6.2.16.1 on embedded linux low perfomance of app_queue |
ASTERISK-17458: Deadlocks when using pthread timer |
ASTERISK-17459: [patch] dont get early media on outgoing calls |
ASTERISK-17460: Crash in ast_frdup |
ASTERISK-17461: DISA allow hangup feature doesn't seem to work |
ASTERISK-17462: Upon using VOLUME() in a dialplan, DTMF tones can get eaten |
ASTERISK-17463: SkypeChatSend with ActionID |
ASTERISK-17464: [patch] sig_pri_new_ast_channel should return NULL when new_ast_channel fails to create a channel |
ASTERISK-17465: Security Vulnerability: AMI access to SHELL function only seems to need CALL Privilege, should be SYSTEM |
ASTERISK-17466: Asterisk Ignores 'remotesecret' parameter |
ASTERISK-17467: external moh is blocked when using dahdi timer |
ASTERISK-17468: [patch] Spectralink 8020 phone will not register |
ASTERISK-17469: ast_string_field_set(clone, name, zombn) |
ASTERISK-17470: [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. |
ASTERISK-17471: "(null)" is printed when there is no peer adress in iax2 show peers |
ASTERISK-17472: [patch] allow storing TCP peers in ast db & alllow rtupdate=no to use them |
ASTERISK-17473: [patch] appdocsxml.dtd updates and general xmldoc updates |
ASTERISK-17474: when using mpg123 as a streaming MOH source, issuing 'moh reload' from CLI causes stream to die |
ASTERISK-17475: CNG or T.38 invite detected while call is bridged causes dialplan execution to stop after ReceiveFax |
ASTERISK-17476: S4A corrupted license |
ASTERISK-17477: Deadlock with blind transfer of SIP call |
ASTERISK-17478: [patch] T.38 state should be unavailable when it is rejected |
ASTERISK-17479: Error obtaining mutex in channel.c |
ASTERISK-17480: [patch] app_voicemail creates "general" mailbox from users.conf |
ASTERISK-17481: Unable to negotiate codec |
ASTERISK-17482: Can't provide secure audio requested in SDP offer |
ASTERISK-17483: Asterisk-R2 outgoing calls without CallerID |
ASTERISK-17484: No moh when call parked |
ASTERISK-17485: [patch] debian init script not lsb compliant |
ASTERISK-17486: SIPshowpeer ends with \r\n\r\n instead of \r\n |
ASTERISK-17487: [patch] Large number of active sip dialogs PUBLISH in the output "sip show channels". |
ASTERISK-17488: no native bridging when more than one crypto offer in SRTP |
ASTERISK-17489: Choppy audio into ConfBridge from IAX call when jitterbuffer=yes |
ASTERISK-17490: [patch] ast_dsp_process logs a error about inband DTMF on faxdetect shut it up |
ASTERISK-17491: CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything |
ASTERISK-17492: Crash in res_phoneprov on reload |
ASTERISK-17493: [patch] dsp.c sends multiple DTMF key events up to applications |
ASTERISK-17494: PickupChan doesn't pick moultiple arguments |
ASTERISK-17495: Deadlock using pickup extension |
ASTERISK-17496: [patch] Small leak in app_externalivr |
ASTERISK-17497: [patch] AELsub() for calling routines that will remain stable between internal changes |
ASTERISK-17498: Asterisk Creates Core Dump on 'core stop gracefully' |
ASTERISK-17499: asterisk crash when i try to change my name or my welcome message |
ASTERISK-17500: chan_h323.c: Unable to create RTP session: Address family not supported by protocol |
ASTERISK-17501: [patch] PATCH: Fixes to MySQL sql files |
ASTERISK-17502: Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x |
ASTERISK-17503: Asterisk ignores reInvite |
ASTERISK-17504: Deadlock using blind transfer |
ASTERISK-17505: Announced transfert with Aastra not works |
ASTERISK-17506: [patch] chan_sip.c with a out-f-call message |
ASTERISK-17507: in strcasecmp () from /lib64/libc.so.6 |
ASTERISK-17508: Asterisk is answering with 404 to OPTIONS messages sent by a SIP peer when this peer try to qualify connection. |
ASTERISK-17509: Occasional SIP deadlock in 1.8.4-rc2 |
ASTERISK-17510: [patch] uncached realtime peers are put in peers_by_ip => memory leak |
ASTERISK-17511: adjust system time to cause asterisk timer run wrong |
ASTERISK-17512: Asterisk-deadlocks - followup to #18497 |
ASTERISK-17513: preferred_codec_only=yes disables the video stream from caller to callee |
ASTERISK-17514: preferred_codec_only=yes disables the video stream from caller to callee |
ASTERISK-17515: groupcount or group doesn't "release" channels and group shows channels which doesn't exists |
ASTERISK-17516: Weird -o parameter for flex |
ASTERISK-17517: [patch] Typo on res_calendar.c - "Cartegories" |
ASTERISK-17518: [patch] Wrong uri in the authentication header |
ASTERISK-17519: Recorded calls corrupted (GSM?) |
ASTERISK-17520: [patch] HANGUP is not sent to AGI in time |
ASTERISK-17521: When transferring a call using the REFER command with replaces enabled, Asterisk drops the call. |
ASTERISK-17522: [patch] 'Silence' is truncated in Record() |
ASTERISK-17523: Qualify for static realtime peers does not work |
ASTERISK-17524: No ringback when 2 or more destinations in dial string |
ASTERISK-17525: Higher penalty member called first |
ASTERISK-17526: No Audio When Using Pickup() Application |
ASTERISK-17527: [patch] There is a resource leak in func_odbc when inserting the previous handle not released |
ASTERISK-17528: Lock on Asterisk 1.6.2.17-rc2 |
ASTERISK-17529: SIPAddHeader in dialplan not in SIP INVITE |
ASTERISK-17530: [patch] (Call Completion / SIP) PUBLISH Fails (412 Conditional Request Failed) |
ASTERISK-17531: IAX2 extension accountcode empty |
ASTERISK-17532: Segmentation Fault booting asterisk |
ASTERISK-17533: [patch] ActionID missing into response when action is events |
ASTERISK-17534: [patch] Deadlock: ast_taskprocessor_get and SIP |
ASTERISK-17535: [patch] [regression] Cisco phones do not register |
ASTERISK-17536: Receiving DTMF for Queue trough SIP trunk fails if not inband. |
ASTERISK-17537: Wrong Column Value When Using res_config_pgsql |
ASTERISK-17538: SIP similar user/peer names are not properly identified when authentication and possibly more |
ASTERISK-17539: after picking up a call, phones stop working... |
ASTERISK-17540: SDP origin attribute modified when issuing re-INVITE because of directmedia=yes |
ASTERISK-17541: Calls from VOIP to Dahdi requiring transcoding fail |
ASTERISK-17542: [patch] Configurable phoneprov path |
ASTERISK-17543: Can't spy direct channel while spying in a group |
ASTERISK-17544: error loading module 'luasql.postgres' |
ASTERISK-17545: Conference menu does not wait after playing message |
ASTERISK-17546: Troubles to compile chan_mobile |
ASTERISK-17547: crash on 1.8.4-rc2 with agi + swift |
ASTERISK-17548: [patch] CEL cel_odbc backend fails to insert eventtype into integer or other numeric column |
ASTERISK-17549: Discrepancy among core-sounds-en.txt and sounds in 1.4.20 set |
ASTERISK-17550: RetryDial Music Bug |
ASTERISK-17551: Do not working lua "app.goto" |
ASTERISK-17552: [patch] 'core show locks' should show Thread ID in HEX, then would match up with GDB's backtrace |
ASTERISK-17553: [patch] Dead code - ast_FD_SETSIZE |
ASTERISK-17554: [patch] Duplicated info in 'rtpqos' description |
ASTERISK-17555: [patch] Remove extra quote in indications.conf |
ASTERISK-17556: Segfault while setting options of a technology driver (1.8.2) |
ASTERISK-17557: Followme not giving the callerid |
ASTERISK-17558: Getting segfault while reloading second time. |
ASTERISK-17559: [patch] chan_misdn segfaults when DEBUG_THREADS is enabled 1.8.4-rc4 |
ASTERISK-17560: Segfault in what seems to be related to RTCP (rtp.c) |
ASTERISK-17561: res_json - dialplan json utilities submitted for review |
ASTERISK-17562: [patch] res_memcached - access to memcached datastores from the dialplan |
ASTERISK-17563: Error after spawn |
ASTERISK-17564: [patch] Timezone variables are corrupted. |
ASTERISK-17565: Call transfer Avaya - Asterisk ooh323 one way audio |
ASTERISK-17566: Unitialized single_binding_found |
ASTERISK-17567: Calls created with Originate will timeout and hangup after 30 seconds |
ASTERISK-17568: crash in musiconhold, using external mpg123 |
ASTERISK-17569: [patch] Can't return to normal ring after distinctive ring on FXS |
ASTERISK-17570: [patch] queue does not handle longest waiting caller first |
ASTERISK-17571: Dubble welcome message (Asterisk Call Manager/1.1) on Asterisk 1.8.3.1 |
ASTERISK-17572: Skinny to Skinny transfer crash |
ASTERISK-17573: Segmentation fault if address can't resolved |
ASTERISK-17574: FollowMe with IAX2 channel LANGUAGE error |
ASTERISK-17575: After reload asterisk stop inserts data to CEL table. |
ASTERISK-17576: issue with Asterisk 1.4.30 and T38 passthrough |
ASTERISK-17577: Wrong Call-ID |
ASTERISK-17578: [patch] DoS through manager interface: no timeout for unauthenticated logins |
ASTERISK-17579: FollowMe not transfter callerid to followme numbers |
ASTERISK-17580: [patch] Voicemail doesn't read/delete messages in one case |
ASTERISK-17581: rtpkeepalive blocks incoming DTMF periodically |
ASTERISK-17582: DNS SRV - does not work |
ASTERISK-17583: Translation of Asterisk voices to the Basque language? |
ASTERISK-17584: Wrong return from Dial app in macro when also used CONNECTED_LINE_CALLER_SEND_MACRO macro call |
ASTERISK-17585: Fax detection always jump to the fax extension |
ASTERISK-17586: Fax detection always jump to the fax extension |
ASTERISK-17587: Voice mailbox left in a non-functional state |
ASTERISK-17588: Caller ID on TDM410P *UK* PSTN |
ASTERISK-17589: Controlplayback don't have multiple forward/rewind option |
ASTERISK-17590: Busy()/Congestion() apps don't work in chan-dahdi with h-extension and priindication=outofband |
ASTERISK-17591: [patch] Remote bridging of certain IPs causes segfault |
ASTERISK-17592: Call stuck with "music on hold" |
ASTERISK-17593: "voicemail show users" command does not return any results since asterisk 1.6.2.13 |
ASTERISK-17594: problem registering gateway gaoke mg6004 4 fxs to asterisk |
ASTERISK-17595: asterisk crashes once a month |
ASTERISK-17596: Call from '1' to extension '1@default' rejected because extension not found. |
ASTERISK-17597: Asterisk crashes with segfault message two or three times a day |
ASTERISK-17598: IAX2 wont start ring back early media |
ASTERISK-17599: [patch] IAX can select the wrong channel name |
ASTERISK-17600: 1.8.4rc2 ooh323 crash on incoming call |
ASTERISK-17601: Error during configure Script and make |
ASTERISK-17602: Error during configure Script |
ASTERISK-17603: Block Agent friendly scanner |
ASTERISK-17604: Very loud crackling noise or no audio on ISDN channel |
ASTERISK-17605: Transfer Q.SIG info from either E1/T1 or OOH323 to Asterisk |
ASTERISK-17606: no audio on iax calls if client registered. |
ASTERISK-17607: [patch] Conversion of hints to AO2 may changes locking order |
ASTERISK-17608: func_aes.so cannot be loaded if res_crypto / openssl not compiled |
ASTERISK-17609: [patch] extra ast_strlen_zero() check in cel_odbc prevents integer eventtype in database |
ASTERISK-17610: Crash on chan_iax2.so |
ASTERISK-17611: Adding record feature to app_externalivr |
ASTERISK-17612: [patch] Xcode4 doesn't support weak_import |
ASTERISK-17613: [patch] Voicemail messages (ODBC), incorrect sequence, msgnum=0 missing! Messages Over written |
ASTERISK-17614: Asterisk crashes with LOW_MEMORY defined |
ASTERISK-17615: Asterisk crash with IAX |
ASTERISK-17616: [patch] fromuser not respected during OPTIONS message (qualify) |
ASTERISK-17617: Crash when posting a CDR record |
ASTERISK-17618: Local Bridging not working |
ASTERISK-17619: [patch] Space Invaders in cmenuselect only has one level |
ASTERISK-17620: clid field empty in mysql table when using cdr_addon_mysql |
ASTERISK-17621: Concurrent access to same mailbox, can delete unread messages. |
ASTERISK-17622: [patch] [regression] H264 video broken due to MARK bit set for each RTP outgoing packet |
ASTERISK-17623: [patch] Add bridge event for local channel |
ASTERISK-17624: not able to change outgoing call pilot no on E1 pri |
ASTERISK-17625: Allow R/W of pickupgroup channel variable |
ASTERISK-17626: user events not receiving |
ASTERISK-17627: bug |
ASTERISK-17628: "T" option is broken - does not wait for extension to be dialed. |
ASTERISK-17629: new bug |
ASTERISK-17630: [patch] Concatenates uninitialized buffer causes garbage data prior result also may cause crash |
ASTERISK-17631: Peers cannot send valid calls |
ASTERISK-17632: Authentication during registration with provider fails when challenge URI contains domain different from request URI |
ASTERISK-17633: [patch] Chan_local crashes in fixup |
ASTERISK-17634: Whisper seems to be broken. Barge works ok. Listen works ok. |
ASTERISK-17635: [patch] asterisk crashes on unattended transfer |
ASTERISK-17636: Re-registration of SIP gateway sometimes occurs after the scheduled delay which causes gateway to see asterisk as unregistered |
ASTERISK-17637: target keeps on ringing on background |
ASTERISK-17638: [patch] CEL Logging to MySQL |
ASTERISK-17639: [patch] SIPAddHeader should modify REFER message originated by Transfer application |
ASTERISK-17640: [patch] Enabled dynamic features swallow DTMF codes (even if no match) |
ASTERISK-17641: Conferences rooms don't ask password |
ASTERISK-17642: [patch] ISDN party subaddress odd_even_indicator inconsitency / undocumented |
ASTERISK-17643: [patch] Column names should be escaped |
ASTERISK-17644: Transfer to Parking Lot no longer plays back parking lot number when using line key on SPA 962, SPA 932 |
ASTERISK-17645: Memory leak in utils.c |
ASTERISK-17646: Asterisk 1.8.3 crash |
ASTERISK-17647: Asterisk crashes with segfault |
ASTERISK-17648: Asterisk crash when it lost connection to Jabber Server |
ASTERISK-17649: crash in ast_frdup with oversized udptl frame |
ASTERISK-17650: Remote-Party ID not added when CALLERID(num)=<empty> |
ASTERISK-17651: Hangup when ChanSpy with rtp p2p channels |
ASTERISK-17652: permit/deny for hostnames |
ASTERISK-17653: [patch] Add 'description' field for SIP peers |
ASTERISK-17654: Invalid Argument errors reported starting asterisk on Darwin |
ASTERISK-17655: pri lockup. no futher calls able to be taken |
ASTERISK-17656: Music On Hold not working |
ASTERISK-17657: [patch] Improve debug of ast_hangup |
ASTERISK-17658: [patch] [regression] Blind transfers from queue are not logged into queue_log correctly |
ASTERISK-17659: [patch] "sip show settings" shows wrong values for jitter buffer settings |
ASTERISK-17660: [patch] Change seconds to milliseconds in ast_verb line |
ASTERISK-17661: [patch] Remove a jabber account |
ASTERISK-17662: "core show channels verbose" and Manager command "coreshowchannels" or "status" not being sorted |
ASTERISK-17663: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3e3b90 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument |
ASTERISK-17664: [patch] FullyBooted event received even when events are disabled |
ASTERISK-17665: [patch] 'voicemail show users for default' with a lot of entries in realtime voicemail cores asterisk |
ASTERISK-17666: [patch] Deadlock in ast_remove_hint Race with ao2_callback |
ASTERISK-17667: Calls getting stuck when dialing *8 |
ASTERISK-17668: [patch] fix detection of openssl 1.0 |
ASTERISK-17669: Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice |
ASTERISK-17670: [patch] tcptls.c fails building with no SSL2 support |
ASTERISK-17671: Busy not being detected on Wildcard TDM410 4-port analog card with HEC |
ASTERISK-17672: Asterisk Does Not Support Delayed Offer - RFC 3264 |
ASTERISK-17673: Fax dimensions are wrong on rcvd faxes |
ASTERISK-17674: T38 termination does not work anymore with Eutelia and Asterisk 1.8 |
ASTERISK-17675: MWI not working for most ATAs |
ASTERISK-17676: audio dropped on attended transfer if first call uses g722 |
ASTERISK-17677: SIP faxdetect does not work when using g729 codec |
ASTERISK-17678: [patch] Image media line with port set to 0 is considered active |
ASTERISK-17679: T38 passthrough does not work with Zoiper zoftphone and Eutelia provider (both Origination and Termination does work) |
ASTERISK-17680: rtp.c:2438 rtp_socket: Unable to allocate RTCP socket: Too many open files |
ASTERISK-17681: Deadlock between rtp_engine.c (ast_rtp_instance_early_bridge) and chan_sip (handle_incoming) |
ASTERISK-17682: [patch] [regression] "sip prune" does not clean the relevant peer objects -> memleak |
ASTERISK-17683: Console flooding caused by bad remote SIP peer |
ASTERISK-17684: Call got Hung up while transfer by logged in agent in call center |
ASTERISK-17685: [regression] When using extenpatternmatchnew=yes, dialplan-based callerid fails using forward slash |
ASTERISK-17686: [patch] Deadlock ao2_callback / ast_write / handle_incoming |
ASTERISK-17687: Reparking a call when using multiple parking lots incorrectly goes to the default parking lot |
ASTERISK-17688: [patch] segfault res_musiconhold.so when called party puts call on hold |
ASTERISK-17689: Duplicate columns retireved for CEL/PostgreSQL & multiple schemas |
ASTERISK-17690: [patch] Crash while transfering a call during DTMF feature timeout. |
ASTERISK-17691: [patch] Start migrating unsupported modules to separate unsupported/ directory |
ASTERISK-17692: In Debian squeeze, make menuselect fails |
ASTERISK-17693: Asterisk Crash many times a day when receiving calls from Dahdi to sip |
ASTERISK-17694: Originate via AMI does not work. |
ASTERISK-17695: 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them |
ASTERISK-17696: international characters in CALLERID(name) should not be escape encoded (for legacy hardware) |
ASTERISK-17697: Deadlocks on 1.6.2.17.2 |
ASTERISK-17698: Astrisk crash when started: Program terminated with signal 11, Segmentation fault. |
ASTERISK-17699: [patch] G.722 support in Google Talk channel |
ASTERISK-17700: Asterisk 1.8.x Skips DTMF Digits on a First DAHDI Initiated Call |
ASTERISK-17701: dahdi drops calls & Unrecognized garbage 'Reserved' in WCTDM/4/2 |
ASTERISK-17702: Crash in chan_sip |
ASTERISK-17703: [patch] 'e' special extension fails to trigger in at least two cases |
ASTERISK-17704: Asterisk Crash when Realtime LDAP extensions not found |
ASTERISK-17705: [patch] One media frame dropped in early media |
ASTERISK-17706: CEL does not logs applications |
ASTERISK-17707: [patch] Unclear code in app_dial.c |
ASTERISK-17708: [patch] Manager event for early media |
ASTERISK-17709: [patch] appdata truncated |
ASTERISK-17710: [patch] Debian removing support for SSLv2 from openssl |
ASTERISK-17711: [patch] SIP 200 OK rejected because the field 'received' in Via header contains link local IPv6 with its scope ! |
ASTERISK-17712: [patch] Asterisk CLI command 'dialplan save' not available |
ASTERISK-17713: [patch] Add new option to meetme to close conference with one remaining user |
ASTERISK-17714: [patch] app_voicemail.c does not compile in 1.4 branch |
ASTERISK-17715: Problems with joinempty and leavewhenempty. |
ASTERISK-17716: Asterisk tries to send an INVITE even when the DNS lookup for the host fails |
ASTERISK-17717: Doing a "channel originate ..." and getting back a 404 leaves the ast_channel up until the sip transaction times out |
ASTERISK-17718: Insecure option in Realtime does not work |
ASTERISK-17719: SIP TLS certificates should be verified according to RFC 5922 |
ASTERISK-17720: Certificate errors don't result in immediate termination of an outbound call |
ASTERISK-17721: Incoming SRTP calls that specify a key lifetime fail |
ASTERISK-17722: SIP SRV lookups for registration discard the port when dnsmgr disabled (the default) |
ASTERISK-17723: Transfer is being declined when global allowtransfer=no regardless of peer settings |
ASTERISK-17724: Voicemail messages are 0-byte when Hangup without # when cache_record_files=yes |
ASTERISK-17725: directmedia or reinvite not working when calling extension that's located an a different asterisk node |
ASTERISK-17726: [patch] Add MixMonitor and StopMixMonitor AMI actions |
ASTERISK-17727: [patch] TLS doesn't get all certificate chain |
ASTERISK-17728: AMR-NB codec. |
ASTERISK-17729: Command "Show Channels" reports incorrect number of active calls in comparison to active channels |
ASTERISK-17730: [regression] moh custom mode with madplay broken with patch from 0016744 |
ASTERISK-17731: Command "Show Channels" reports incorrect number of active calls in comparison to active channels |
ASTERISK-17732: [patch] With multiple queues & agents, calls with the longest waiting time don't always get handled first |
ASTERISK-17733: dial(dahdi/) crashed asterisk |
ASTERISK-17734: Dialing dial(sip/) with emtpy parameter gives strage results |
ASTERISK-17735: Hangup function not sending correct SIP cause code |
ASTERISK-17736: Add bleepin and bleepout ConfBridge join sounds before 1.10 release |
ASTERISK-17737: AMI Action: Queuestatus + asterisk -rx "queue show" = no more calls sent to agents |
ASTERISK-17738: [patch] Multiple parking lots parkedcall/transfers/reparking/hangup/recording no handled properly |
ASTERISK-17739: possible deadlock in WaitExten() |
ASTERISK-17740: Deadlock in handle_request_bye() and mutex error in handle_incoming() |
ASTERISK-17741: Nationalprefix chan_dahdi option ignored |
ASTERISK-17742: ulaw/g722 Transcode Issue |
ASTERISK-17743: 1.8.4-rc2 fails when loading Skype |
ASTERISK-17744: RTP timestamp skewed after call transfer or call unhold |
ASTERISK-17745: Error coming while make install command |
ASTERISK-17746: QueuePickup |
ASTERISK-17747: [patch] check_bridge(): misplaced ast_mutex_unlock |
ASTERISK-17748: pickup beep choppy. debug log shows "Resource temporarily unavailable" |
ASTERISK-17749: [patch] MeetMeChannelAdmin over AMI |
ASTERISK-17750: DTMF transmission does not meet the recommendations of the rfc2833(4733). |
ASTERISK-17751: [patch] Random crashes (NULL reference) at res_odbc.c:1358 |
ASTERISK-17752: (not so)large astDB key values cause asterisk to take up a lot of memory |
ASTERISK-17753: [patch] [regression] Asterisk drops sip messages and/or response codes if SIP/TLS is used |
ASTERISK-17754: [patch] OOH323 does not do full T.38 Handshaking / Fax Detection is limited |
ASTERISK-17755: DTMF ( RFC2833 or SIP INFO ) sent incorrectly when bridged to a DAHDI channel on remote server. |
ASTERISK-17756: [patch] Asterisk crashes with a segfault if current host cannot be resolved via DNS. |
ASTERISK-17757: [patch] dahdi_hangup() doesnt clean up / hang up the channel correctly/fully |
ASTERISK-17758: How to get hold time to caller's in queue??? |
ASTERISK-17759: When calling myself with a SIP TRUNK with ITSP provider, incoming call is considered diverted and channel converted to Local/ |
ASTERISK-17760: [patch] deadlock in chan_sip |
ASTERISK-17761: [patch] [regression] segfault in _sip_tcp_helper_thread() caused by bad merge in r314628 |
ASTERISK-17762: [patch] Confusing directory vestige /var/log/asterisk/cel-csv/ |
ASTERISK-17763: sip.conf.sample incorrectly describes types (peer/user/friend) |
ASTERISK-17764: [patch] When SIP caller does not offer video, adding video drops the call |
ASTERISK-17765: Can't provide secure audio requested in SDP offer |
ASTERISK-17766: Outbound DTMF issues |
ASTERISK-17767: Channel Group exists for nonexistent channel (didn't clear out properly) |
ASTERISK-17768: [patch] MaxMsg Quota Only Enforced on INBOX Messages for Filesystem Based Voice Mail Boxes |
ASTERISK-17769: [patch] sometimes ntohs in place of htons |
ASTERISK-17770: Asterisk crashes in ast_cel_report_event on hangup when obtaining invalid bridged channel |
ASTERISK-17771: [patch] switching From-address mid-register breaks channel variables |
ASTERISK-17772: mutex 'current_dest_chan' freed more times than we've locked! on bridge() from dialplan |
ASTERISK-17773: Asterisk 1.6.0.22 crash |
ASTERISK-17774: Using "loose" option for "joinempty" and/or "leavewhenempty" settings causes severe Asterisk issues |
ASTERISK-17775: Deadlock with Dahdi |
ASTERISK-17776: No CDR record is generated if caller hangs up while in Queue and members are busy. Reproduced in 1.8 and 1.6 as well. |
ASTERISK-17777: Voicemail cuts off at 60 seconds regardless of config settings - the same problem |
ASTERISK-17778: configure has syntax error after r316006 committed |
ASTERISK-17779: Error loading module 'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open shared object file: No such file or |
ASTERISK-17780: Hostname does not resolve when using realtime SIP |
ASTERISK-17781: warning-chan-sip-c-sip-xmit-returned-2-interrupted-system-call |
ASTERISK-17782: [patch] bug in contrib/scripts/safe_asterisk |
ASTERISK-17783: Call declined when rtupdate=no |
ASTERISK-17784: [patch] Framehook Segfaults on indicate |
ASTERISK-17785: [patch] Originate action generates two error responses |
ASTERISK-17786: Fullybooted event is always sent after login |
ASTERISK-17787: SIP registration fails in parsing |
ASTERISK-17788: Memleak with every incoming SUBSCRIBE. |
ASTERISK-17789: [patch] Peer address always overwritten by contact address |
ASTERISK-17790: CPU to 100% with many call in MusicOnHold |
ASTERISK-17791: [patch] realtime peer update fails, because "" is not a valide int value for lastms |
ASTERISK-17792: Posible memleak in realtime_peer |
ASTERISK-17793: [patch] sip_set_rtp_peer sometimes returns too early |
ASTERISK-17794: pbx_spool does not detect callfiles over NFS |
ASTERISK-17795: [patch] Monitor files get incorrectly padded during transfer/masq |
ASTERISK-17796: Segmentation fault if not specified SIP extension at Dial |
ASTERISK-17797: Coding error in find_subchannel_and_lock() |
ASTERISK-17798: pbx_lua hangs on faxdetect |
ASTERISK-17799: AEL reload causes loss of control in a macro |
ASTERISK-17800: Phones Ring Forever When Termination Signal Comes Shortly After Initiating Signal |
ASTERISK-17801: Adding the Move to Front Hash functionality |
ASTERISK-17802: SIP stack stops working if a Dial command if forwarded by a SIP physical phone |
ASTERISK-17803: [patch] Add sample offset parameter to CONTROL STREAM FILE |
ASTERISK-17804: [patch] Separate mailbox and mailboxname |
ASTERISK-17805: [patch] prependfile in __ast_play_and_record |
ASTERISK-17806: [patch] announce field in queue_ent is not long enough |
ASTERISK-17807: [patch] Voicemail normalization |
ASTERISK-17808: [patch] Unregister a realtime moh class |
ASTERISK-17809: [patch] Append Asterisk UniqueId to Monitor Filenames |
ASTERISK-17810: [patch] Separate directories for running and core dumps |
ASTERISK-17811: [patch] find_table does not take pgsql_lock before using pgsqlConn |
ASTERISK-17812: [patch] Missing calls to PQclear |
ASTERISK-17813: [patch] No inband progress on PRI_EVENT_RINGING even if inband flag set |
ASTERISK-17814: When a call is blind-transfered to another extension, the CLI wrongly display the transfered call as been hung up. |
ASTERISK-17815: [patch] [regression] SRV lookup attempted for peers listed by IP address |
ASTERISK-17816: Asterisk locks up at times and calls don't go through |
ASTERISK-17817: [patch] Asterisk res_odbc causes SEGFAULT's if unable to connect to server. |
ASTERISK-17818: [patch] Framehooks ast_indicate_data Incorectly uses "read" this should be "write" |
ASTERISK-17819: error on SIP INVITE when host's external IP changes |
ASTERISK-17820: Random crash with RES_ODBC related to SQLSetConnectAttr(). |
ASTERISK-17821: Def. Username field is incorrect |
ASTERISK-17822: Can't disable endcall. |
ASTERISK-17823: [patch] Crash when using hagi and no servers are available |
ASTERISK-17824: [patch] Unable to pickup ringing DAHDI call, as when ringing state is reported as DIALLING |
ASTERISK-17825: [patch] dynamic thread exits with joinable state, which leaves resource of thread (250kb stack per thread etc.) unreleased |
ASTERISK-17826: [patch] 3 examples of loss of CDR data |
ASTERISK-17827: [patch] Manager eventfilter blacklisting does not filter blacklisted events |
ASTERISK-17828: Add an argument to specify the number of digits (not hard coded) |
ASTERISK-17829: Duplicate SIP 180 / Ringing Messages in SIP trunk ingress |
ASTERISK-17830: [regression] Cisco phones do not register |
ASTERISK-17831: Asterisk crash when comunicate with Nuace |
ASTERISK-17832: [regression] Deadlock in chan_sip |
ASTERISK-17833: Hold music lost after attended transfer |
ASTERISK-17834: [patch] pri early media won't ring |
ASTERISK-17835: procoess is stopped in middle |
ASTERISK-17836: Reg:Multiple RegistrationOfUserFromSipClient |
ASTERISK-17837: extconfig.conf - Maximum Include level (1) exceeded |
ASTERISK-17838: [patch] remote authenticated asterisk DoS (crash) for tcp-sip clients |
ASTERISK-17839: [patch] app_privacy arguments reference invalid "options" option, interferes with "context" option |
ASTERISK-17840: [patch] Deadlock on transferring |
ASTERISK-17841: Notify contact and via header port is 0 |
ASTERISK-17842: [patch] Building BETTER_BACKTRACES on CentOS 5 |
ASTERISK-17843: change BOOL to PBoolean |
ASTERISK-17844: [patch] Not correct return from ooh323_indicate(..) function for interoperating with res_fax (in particular) |
ASTERISK-17845: [patch] Invite with session description that supports both SRTP(SAVP) and RTP(AVP) fails if asterisk is not configured to use SR |
ASTERISK-17846: [crash] SegFault / TCP enabled in _sip_tcp_helper_thread |
ASTERISK-17847: Dialing on nonexistent SIP-peer causes __sip_xmit errors |
ASTERISK-17848: Asterisk 1.8 dead with 100% CPU |
ASTERISK-17849: [patch] inbound call from google voice fail |
ASTERISK-17850: inverse / incorrect behavior for CLI / console logging of DTMF |
ASTERISK-17851: module unload chan_iax2.so will crash asterisk |
ASTERISK-17852: [patch] ast_tcptls_server_start fails second attempt to bind |
ASTERISK-17853: Asterisk Crashing |
ASTERISK-17854: [patch] wrong description of queuestrategy leastrecent |
ASTERISK-17855: Endpoint call forwarding fails with congestion |
ASTERISK-17856: tlsverifyclient option not working |
ASTERISK-17857: [patch] CCSS Crash introduced in new SVN [318867] |
ASTERISK-17858: if ODBC can´t connect asterisk crashes |
ASTERISK-17859: chan_sip crashes since changes in rev. 318549 |
ASTERISK-17860: [patch] Auto detect NAT if the same device can be used in several natted/unnated scenarios |
ASTERISK-17861: [regression] Big Endian breakage (regression since gcc 4.6 fix) |
ASTERISK-17862: Crash in cel_pgsql |
ASTERISK-17863: Backported DEVICE_STATE function code still mentions old name DEVSTATE in some logging statements |
ASTERISK-17864: Called channel stays in app_dial_gosub_virtual_context and can't transfer call properly. |
ASTERISK-17865: app_queue INVITE-ing unregistred SIP |
ASTERISK-17866: [patch] MWI last-msgs-sent is mis-reported |
ASTERISK-17867: [patch] Random entire asterisk deadlock when use builtin_atxfer when use res_timing_timerfd module |
ASTERISK-17868: deadlock while reload |
ASTERISK-17869: [patch] MOH is not played when quiet=no |
ASTERISK-17870: Cannot retrieve parked calls |
ASTERISK-17871: Asterisk always deadlocks in queue after upgrading to 1.8.3&1.8.4 |
ASTERISK-17872: Asterisk 1.6.2.17.2 randomly chrashes after 1 to 3 days of normal operation |
ASTERISK-17873: [patch] Crash when using PickupChan |
ASTERISK-17874: [patch] [regression] Revision 315643 app_dial breaks ring groups |
ASTERISK-17875: Asterisk "forgets" default music on hold files |
ASTERISK-17876: Infinite loop after "queue show" with realtime Postgreql queues |
ASTERISK-17877: [patch] Last character of XML data gets cut off, in mxml interface |
ASTERISK-17878: [patch] Generating output using cdr_custom backend is corrupt because of pbx.c |
ASTERISK-17879: FastAGI looping when an error happens on fgets |
ASTERISK-17880: [patch] [regression] SIP peers unregistered when asterisk restarts |
ASTERISK-17881: Asterisk Crashes on CLI reload, moh reload, or randomly itself |
ASTERISK-17882: alarmreceiver doesnt work |
ASTERISK-17883: SIP CANCEL is broken when phone is not registered to asterisk (sip friend and/or sip user) |
ASTERISK-17884: Registration info lost after "sip reload" for TCP endpoints |
ASTERISK-17885: Local Channels not passing DTMF Tones properly |
ASTERISK-17886: autodomain in sip.conf should be smart enough to detect alternate port from udpbindaddr or similar setting |
ASTERISK-17887: SendFax does not work probably as it keeps sending INVITES |
ASTERISK-17888: [patch] In Manager Interface, SIP registry event does not show username on Status: Registered |
ASTERISK-17889: ExternalIVR Crash |
ASTERISK-17890: Signal 11 segmentation fault 0x0809495e in ast_do_masquerade |
ASTERISK-17891: [patch] Deadlock in chan_sip.c handle_incoming() |
ASTERISK-17892: [patch] Crash on load of cel_odbc |
ASTERISK-17893: Segfault in chan_sip probably related to media formats/codecs |
ASTERISK-17894: Asterisk to Asterisk multiple registration goes to "username mismatch" |
ASTERISK-17895: [patch] chan_sip encryption attempt srtp / set auth taglen |
ASTERISK-17896: [patch] meetme cli cmd completion leaves conferences mutex locked |
ASTERISK-17897: [patch] Call shows on hold after attended transfer with a Polycom phone |
ASTERISK-17898: Call Sent to invlaid Macro Asterisk crashes |
ASTERISK-17899: Handle crypto lifetime in SDES-SRTP negotiation |
ASTERISK-17900: Lost SIP registration from cache/database after restart asterisk |
ASTERISK-17901: [patch] Timout or error on INFO or MESSAGE transaction causes call to be lost |
ASTERISK-17902: [patch] restrict nativeformat to negociated codecs |
ASTERISK-17903: unable to unload res_jabber if connetion to server can not established |
ASTERISK-17904: [patch] autoconf check for iconv does not work for GNU libiconv |
ASTERISK-17905: [patch] Crash in netsock when attempting to resolve blank sip uri |
ASTERISK-17906: no musiconhold when using RetryDial |
ASTERISK-17907: [patch] chanspy enforced mode lacks a channel_unlock |
ASTERISK-17908: [patch] Deadlock ast_channel_cmp_cb |
ASTERISK-17909: [patch] Crash in chan_sip -- sip_setoption() |
ASTERISK-17910: Sip NOTIFY does not populate polycom Callerid FROM |
ASTERISK-17911: [patch] Ability to dynamically add filters to an ami session |
ASTERISK-17912: Add option to app_read to allow for capturing # at end of input |
ASTERISK-17913: Asterisk crashed on app_queue random |
ASTERISK-17914: [regression] Change 310888 (Don't delay DTMF) broke DYNAMIC_FEATURES |
ASTERISK-17915: [patch] Reboot Snom phones with "sip notify snom-reboot" |
ASTERISK-17916: First DAHDI line transferred into ConfBridge hung up when second DAHDI line transferred into conference |
ASTERISK-17917: [patch] app_dial may double free a channel datastore |
ASTERISK-17918: Context jump use causes MeetMe to hard-hangup (Context scope bug) |
ASTERISK-17919: [patch] OOH323 Unexpectedly Drops Incoming Forwarded Calls in 15 Seconds |
ASTERISK-17920: [patch] SLA can send a HOLD hint when trunk is not held |
ASTERISK-17921: Detecting Caller id before first ring |
ASTERISK-17922: Stuck waiting to execute the sip scheduler queue forever |
ASTERISK-17923: [patch] New manager option enabledevents |
ASTERISK-17924: Message ID does not match trid field in message structure |
ASTERISK-17925: CPU spike if hang up during getting message "You are currently the only person in this conference" |
ASTERISK-17926: The retrans_pkt function can corrupt the message list in the gateway structure |
ASTERISK-17927: [patch] Invalid read and null pointer deref on asterisk shutdown |
ASTERISK-17928: trixbox v2.8.0.4 ©2008 - System Status Version: 2.6.2.5 |
ASTERISK-17929: Attended transfer broken - 481 Call/Transaction Does Not Exist |
ASTERISK-17930: Attended transfer - transfering phone left connected |
ASTERISK-17931: [patch] Asterisk do not respect order of queue members |
ASTERISK-17932: Unable to send or receive calls thru DAHDI |
ASTERISK-17933: Random crashes |
ASTERISK-17934: Asterisk SIP Freezes |
ASTERISK-17935: [patch] Parking lot hint and mohclass |
ASTERISK-17936: [patch] Fail to play video prompts |
ASTERISK-17937: [patch] improvment on SendText() to use realtime text if negociated |
ASTERISK-17938: [patch] Segmentation faults |
ASTERISK-17939: [patch] used auth= parameter freed during sip reload => crash |
ASTERISK-17940: [patch] Peer not recorded on CEL_BRIDGE_START and CEL_BRIDGE_END |
ASTERISK-17941: add queue-minute option on queues.conf example file |
ASTERISK-17942: Missing session-expires from Asterisk to the phone. Asterisk not working according to RFC |
ASTERISK-17943: Incorrect owner/group log files |
ASTERISK-17944: Display names with '\' characters near quotes make Asterisk fail to parse the quotation marks correctly. |
ASTERISK-17945: [patch] NOTICE message says what wasn't reachable, but not who couldn't reach it |
ASTERISK-17946: AST_STRING_FIELD new pointer overlap the last string on MIPS |
ASTERISK-17947: [patch] Add SLA support to chan_skinny |
ASTERISK-17948: [patch] missing ast_channel_lock() in func_odbc.c function acf_fetch() |
ASTERISK-17949: Asterisk Hangs after receiving SIP response 302 "Moved Temporarily" from Yealink T22P Phone |
ASTERISK-17950: [patch] Deadlock in queue handling |
ASTERISK-17951: clid field empty in mysql table when using cdr_addon_mysql |
ASTERISK-17952: RTP Timestamp jump on marker packet |
ASTERISK-17953: [patch] CCSS: Sending a NOTIFY without the Subscription-State header |
ASTERISK-17954: Accountcode and Peeraccount is not set correctly |
ASTERISK-17955: Segmentation fault with LibSS7 |
ASTERISK-17956: [patch] Call pickup deadlock in ast_do_masquerade |
ASTERISK-17957: [patch] Default mysql socket selection code makes module config load to fail |
ASTERISK-17958: [patch] The debian init script is still not LSB compliant, #0018896 should not have been closed |
ASTERISK-17959: Buffer overflow in custom_prepare |
ASTERISK-17960: When using chanspy on a monitored channel the file is never closed. |
ASTERISK-17961: crash with Pickup and PickupChan |
ASTERISK-17962: a patch to fix bug 18130 breaks sequential emulated rfc2833 dtmf |
ASTERISK-17963: crash when on disconnect |
ASTERISK-17964: Possible memory leak in chan_sip.c |
ASTERISK-17966: SEGMENTATION FAULTS/CRASH |
ASTERISK-17967: Asterisk locks on transfer |
ASTERISK-17968: chan_dahdi - core dump at Asterisk startup if red alarm on PRI (1.8 SVN) |
ASTERISK-17969: Asterisk 1.8.2.3 crashes when dialling from IAX2 to IAX2 |
ASTERISK-17970: crash on core restart when convenient |
ASTERISK-17971: awk syntax error in iLBC contrib for solaris |
ASTERISK-17973: Possible error in variable declaration in /include/asterisk/logger.h |
ASTERISK-17974: DTMF events are not logged to console due to possible variable init error |
ASTERISK-17975: Using ChanSpy getting repeated WARNING[9901]: chan_sip.c:6083 |
ASTERISK-17976: bindaddr only sets udp port |
ASTERISK-17977: loop for playback crashes |
ASTERISK-17978: XMPP Text Messages Sent To Dialplan |
ASTERISK-17980: Voicemail files "skip" slightly |
ASTERISK-17981: Crash in ast_db_put |
ASTERISK-17982: We are not able to make inbound calls and Outbound Caller Id is not displaying in the phone. |
ASTERISK-17983: Extenstion sort and n priority |
ASTERISK-17984: Pri line error cause 44 |
ASTERISK-17985: Syntax errors in dialplan do not display the file name |
ASTERISK-17986: Config inheritance doesn't work with ConfBridge() menu definitions |
ASTERISK-17988: MOH for only user not working with ConfBridge() |
ASTERISK-17990: Architectural problem with FSK callerid on analog lines |
ASTERISK-17992: chan_mgcp segmentation fault |
ASTERISK-17993: Inbound Google Chat calls fail |
ASTERISK-17994: Regression - GROUP_COUNT value not decremented on channel hangup |
ASTERISK-17995: Asterisk crash with signal 8, Arithmetic exception in Dial application by Limit option with warning frequency equals to zero |
ASTERISK-17996: timing dahdi priority |
ASTERISK-17997: When subscribing mwi to an unsolicited mailbox the first notification is incorrect |
ASTERISK-17998: Hangup with active AGI connection does not work |
ASTERISK-17999: Documentation tweak for AGI Hangup |